With the rapid development of the Voice Over IP, VOIP, service, application occasions for services such as calls across the Circuit Switched, CS, Domain and the Packet Switched, PS, Domain as well as Peer-to-Peer, P2P, calls etc. also become more and more.
Business scenarios, where it is needed to transfer a dual-party conversation into a multi-party conference, usually occur in the process of people being in a VOIP conversation. There are several existing implementation modes as follows: one mode is to implement one local conference room for a multi-party conversation at a client, thus implementing a business scenario where a dual-party conversation is transferred into a multi-party conversation. Another mode is to release an existing dual-party call, create one new conference room, and pull related conference participants into the conference room, so as to manually implement transferring both types of calls into a conference.
However, for the first mode, under the scenario where sounds are mixed locally at the terminal, each time an anchorperson adds a conversation of one party, one path of local bandwidth will be added, which has higher requirements on the network locally at the terminal; and in addition, each time one path of terminal is added, one path of sound mixing codec processing is needed, which requires the terminal to have higher software processing capacity for a VOIP application with higher requirements on the immediacy, thus limiting the occasions for which the implementation mode for mixing sounds at the terminal is suitable. The last of the above modes needs to cut off the current conversation and re-create a mode of inviting the opposite party and a third party to participate in the conference, which results in interruption of the current conversation and brings extreme inconvenience to the user.